//
// Copyright (c) 2013-2021 Winlin
//
// SPDX-License-Identifier: MIT
//

'use strict';

function SrsError(name, message) {
    this.name = name;
    this.message = message;
    this.stack = new Error().stack;
}
SrsError.prototype = Object.create(Error.prototype);
SrsError.prototype.constructor = SrsError;

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher.
function SrsRtcPublisherAsync() {
    var self = {};

    const isMobile = () => {
        let isMobile = false;
        if (navigator.userAgent) {
            const mobileKeyword = ['mobile', 'iphone', 'android'];
            mobileKeyword.forEach((keyword) => {
                if (navigator.userAgent.toLocaleLowerCase().includes(keyword)) {
                    isMobile = true;
                }
            });
        }
        return isMobile;
    };

    const envPar = isMobile()
        ? {
              facingMode: { exact: 'environment' },
          }
        : {};

    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
    self.constraints = {
        audio: {
            echoCancellation: true, // 回音消除
            noiseSuppression: true, // 噪声抑制
        },
        video: {
            width: { ideal: 512, max: 912 },
            height: { ideal: 910, max: 1621.3 },
            ...envPar,
        },
    };

    // @see https://github.com/rtcdn/rtcdn-draft
    // @url The WebRTC url to play with, for example:
    //      webrtc://r.ossrs.net/live/livestream
    // or specifies the API port:
    //      webrtc://r.ossrs.net:11985/live/livestream
    // or autostart the publish:
    //      webrtc://r.ossrs.net/live/livestream?autostart=true
    // or change the app from live to myapp:
    //      webrtc://r.ossrs.net:11985/myapp/livestream
    // or change the stream from livestream to mystream:
    //      webrtc://r.ossrs.net:11985/live/mystream
    // or set the api server to myapi.domain.com:
    //      webrtc://myapi.domain.com/live/livestream
    // or set the candidate(eip) of answer:
    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
    // or force to access https API:
    //      webrtc://r.ossrs.net/live/livestream?schema=https
    // or use plaintext, without SRTP:
    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
    // or any other information, will pass-by in the query:
    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
    //      webrtc://r.ossrs.net/live/livestream?token=xxx
    self.publish = async function (url, callback) {
        var conf = self.__internal.prepareUrl(url);
        self.pc.addTransceiver('audio', { direction: 'sendonly' });
        self.pc.addTransceiver('video', { direction: 'sendonly' });

        if (
            !navigator.mediaDevices &&
            window.location.protocol === 'http:' &&
            window.location.hostname !== 'localhost'
        ) {
            throw new SrsError(
                'HttpsRequiredError',
                `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`,
            );
        }
        try {
            var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
            console.log('[srs] getUserMedia suc', stream);
            callback(stream);
        } catch (e) {
            console.log('[srs] getUserMedia fail', e);
            callback(null);
            return null;
        }
        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
        stream.getTracks().forEach(function (track) {
            console.log('[srs]add track', track);
            self.pc && self.pc.addTrack(track);

            // Notify about local track when stream is ok.
            self.ontrack && self.ontrack({ track: track });
        });

        const senders = self.pc.getSenders();
        senders.forEach(async (sender) => {
            if (sender && sender.track.kind === 'video') {
                const params = sender.getParameters();
                params.encodings[0].maxBitrate = 1024 * 1024 * 30;
                await sender.setParameters(params);
            }
        });

        console.log('[srs]createoffer');
        var offer = await self.pc.createOffer();
        await self.pc.setLocalDescription(offer);
        var session = await new Promise(function (resolve, reject) {
            // @see https://github.com/rtcdn/rtcdn-draft
            var data = {
                api: conf.apiUrl,
                tid: conf.tid,
                streamurl: conf.streamUrl,
                clientip: null,
                sdp: offer.sdp,
            };
            console.log('[srs]Generated offer: ', data, conf.apiUrl);

            const xhr = new XMLHttpRequest();
            xhr.onload = function () {
                if (xhr.readyState !== xhr.DONE) return;
                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                const data = JSON.parse(xhr.responseText);
                console.log('[srs]Got answer: ', data);
                return data.code ? reject(xhr) : resolve(data);
            };
            xhr.open('POST', conf.publishPath, true);
            xhr.setRequestHeader('Content-type', 'application/json');
            xhr.send(JSON.stringify(data));
        });
        console.log('[srs]set remote desc', session.sdp);
        await self.pc.setRemoteDescription(new RTCSessionDescription({ type: 'answer', sdp: session.sdp }));
        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';

        return session;
    };

    // Close the publisher.
    self.close = function () {
        self.pc && self.pc.close();
        self.pc = null;
    };

    // The callback when got local stream.
    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
    self.ontrack = function (event) {
        // Add track to stream of SDK.
        self.stream && self.stream.addTrack(event.track);
    };

    // Internal APIs.
    self.__internal = {
        defaultPath: '/rtc/v1/publish/',
        prepareUrl: function (webrtcUrl) {
            var urlObject = self.__internal.parse(webrtcUrl);

            // If user specifies the schema, use it as API schema.
            var schema = urlObject.user_query.schema;
            schema = schema ? schema + ':' : window.location.protocol;

            var port = urlObject.port || 1985;
            if (schema === 'https:') {
                port = urlObject.port || 443;
            }

            // @see https://github.com/rtcdn/rtcdn-draft
            var api = urlObject.user_query.play || self.__internal.defaultPath;
            if (api.lastIndexOf('/') !== api.length - 1) {
                api += '/';
            }

            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
            for (var key in urlObject.user_query) {
                if (key !== 'api' && key !== 'play') {
                    apiUrl += '&' + key + '=' + urlObject.user_query[key];
                }
            }
            // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
            apiUrl = apiUrl.replace(api + '&', api + '?');

            var streamUrl = urlObject.url;

            return {
                apiUrl: apiUrl,
                streamUrl: streamUrl,
                schema: schema,
                urlObject: urlObject,
                port: port,
                publishPath: self.__internal.defaultPath,
                tid: Number(parseInt(new Date().getTime() * Math.random() * 100))
                    .toString(16)
                    .slice(0, 7),
            };
        },
        parse: function (url) {
            // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
            var a = document.createElement('a');
            a.href = url.replace('rtmp://', 'http://').replace('webrtc://', 'http://').replace('rtc://', 'http://');

            var vhost = a.hostname;
            var app = a.pathname.substring(1, a.pathname.lastIndexOf('/'));
            var stream = a.pathname.slice(a.pathname.lastIndexOf('/') + 1);

            // parse the vhost in the params of app, that srs supports.
            app = app.replace('...vhost...', '?vhost=');
            if (app.indexOf('?') >= 0) {
                var params = app.slice(app.indexOf('?'));
                app = app.slice(0, app.indexOf('?'));

                if (params.indexOf('vhost=') > 0) {
                    vhost = params.slice(params.indexOf('vhost=') + 'vhost='.length);
                    if (vhost.indexOf('&') > 0) {
                        vhost = vhost.slice(0, vhost.indexOf('&'));
                    }
                }
            }

            // when vhost equals to server, and server is ip,
            // the vhost is __defaultVhost__
            if (a.hostname === vhost) {
                var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
                if (re.test(a.hostname)) {
                    vhost = '__defaultVhost__';
                }
            }

            // parse the schema
            var schema = 'rtmp';
            if (url.indexOf('://') > 0) {
                schema = url.slice(0, url.indexOf('://'));
            }

            var port = a.port;
            if (!port) {
                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
                    port = url.indexOf(`webrtc://${a.host}:80`) === 0 ? 80 : 443;
                }

                // Guess by schema.
                if (schema === 'http') {
                    port = 80;
                } else if (schema === 'https') {
                    port = 443;
                } else if (schema === 'rtmp') {
                    port = 1935;
                }
            }

            var ret = {
                url: url,
                schema: schema,
                server: a.hostname,
                port: port,
                vhost: vhost,
                app: app,
                stream: stream,
            };
            self.__internal.fill_query(a.search, ret);

            // For webrtc API, we use 443 if page is https, or schema specified it.
            if (!ret.port) {
                if (schema === 'webrtc' || schema === 'rtc') {
                    if (ret.user_query.schema === 'https') {
                        ret.port = 443;
                    } else if (window.location.href.indexOf('https://') === 0) {
                        ret.port = 443;
                    } else {
                        // For WebRTC, SRS use 1985 as default API port.
                        ret.port = 1985;
                    }
                }
            }

            return ret;
        },
        fill_query: function (query_string, obj) {
            // pure user query object.
            obj.user_query = {};

            if (query_string.length === 0) {
                return;
            }

            // split again for angularjs.
            if (query_string.indexOf('?') >= 0) {
                query_string = query_string.split('?')[1];
            }

            var queries = query_string.split('&');
            for (var i = 0; i < queries.length; i++) {
                var elem = queries[i];

                var query = elem.split('=');
                obj[query[0]] = query[1];
                obj.user_query[query[0]] = query[1];
            }

            // alias domain for vhost.
            if (obj.domain) {
                obj.vhost = obj.domain;
            }
        },
    };

    self.pc = new RTCPeerConnection(null);

    // To keep api consistent between player and publisher.
    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
    // @see https://webrtc.org/getting-started/media-devices
    self.stream = new MediaStream();

    return self;
}

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-await-promise based SRS RTC Player.
function SrsRtcPlayerAsync() {
    var self = {};

    // @see https://github.com/rtcdn/rtcdn-draft
    // @url The WebRTC url to play with, for example:
    //      webrtc://r.ossrs.net/live/livestream
    // or specifies the API port:
    //      webrtc://r.ossrs.net:11985/live/livestream
    //      webrtc://r.ossrs.net:80/live/livestream
    // or autostart the play:
    //      webrtc://r.ossrs.net/live/livestream?autostart=true
    // or change the app from live to myapp:
    //      webrtc://r.ossrs.net:11985/myapp/livestream
    // or change the stream from livestream to mystream:
    //      webrtc://r.ossrs.net:11985/live/mystream
    // or set the api server to myapi.domain.com:
    //      webrtc://myapi.domain.com/live/livestream
    // or set the candidate(eip) of answer:
    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
    // or force to access https API:
    //      webrtc://r.ossrs.net/live/livestream?schema=https
    // or use plaintext, without SRTP:
    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
    // or any other information, will pass-by in the query:
    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
    //      webrtc://r.ossrs.net/live/livestream?token=xxx
    self.play = async function (url) {
        var conf = self.__internal.prepareUrl(url);
        self.pc.addTransceiver('audio', { direction: 'recvonly' });
        self.pc.addTransceiver('video', { direction: 'recvonly' });
        //self.pc.addTransceiver("video", {direction: "recvonly"});
        //self.pc.addTransceiver("audio", {direction: "recvonly"});

        var offer = await self.pc.createOffer();
        await self.pc.setLocalDescription(offer);
        var session = await new Promise(function (resolve, reject) {
            // @see https://github.com/rtcdn/rtcdn-draft
            var data = {
                api: conf.apiUrl,
                tid: conf.tid,
                streamurl: conf.streamUrl,
                clientip: null,
                sdp: offer.sdp,
            };
            console.log('Generated offer: ', data);

            const xhr = new XMLHttpRequest();
            xhr.onload = function () {
                if (xhr.readyState !== xhr.DONE) return;
                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                const data = JSON.parse(xhr.responseText);
                console.log('Got answer: ', data);
                return data.code ? reject(xhr) : resolve(data);
            };
            xhr.open('POST', conf.apiUrl, true);
            xhr.setRequestHeader('Content-type', 'application/json');
            xhr.send(JSON.stringify(data));
        });
        await self.pc.setRemoteDescription(new RTCSessionDescription({ type: 'answer', sdp: session.sdp }));
        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';

        return session;
    };

    // Close the player.
    self.close = function () {
        self.pc && self.pc.close();
        self.pc = null;
    };

    // The callback when got remote track.
    // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
    self.ontrack = function (event) {
        // https://webrtc.org/getting-started/remote-streams
        self.stream.addTrack(event.track);
    };

    // Internal APIs.
    self.__internal = {
        defaultPath: '/rtc/v1/play/',
        prepareUrl: function (webrtcUrl) {
            var urlObject = self.__internal.parse(webrtcUrl);

            // If user specifies the schema, use it as API schema.
            var schema = urlObject.user_query.schema;
            schema = schema ? schema + ':' : window.location.protocol;

            var port = urlObject.port || 1985;
            if (schema === 'https:') {
                port = urlObject.port || 443;
            }

            // @see https://github.com/rtcdn/rtcdn-draft
            var api = urlObject.user_query.play || self.__internal.defaultPath;
            if (api.lastIndexOf('/') !== api.length - 1) {
                api += '/';
            }

            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
            for (var key in urlObject.user_query) {
                if (key !== 'api' && key !== 'play') {
                    apiUrl += '&' + key + '=' + urlObject.user_query[key];
                }
            }
            // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
            apiUrl = apiUrl.replace(api + '&', api + '?');

            var streamUrl = urlObject.url;

            return {
                apiUrl: apiUrl,
                streamUrl: streamUrl,
                schema: schema,
                urlObject: urlObject,
                port: port,
                tid: Number(parseInt(new Date().getTime() * Math.random() * 100))
                    .toString(16)
                    .slice(0, 7),
            };
        },
        parse: function (url) {
            // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
            var a = document.createElement('a');
            a.href = url.replace('rtmp://', 'http://').replace('webrtc://', 'http://').replace('rtc://', 'http://');

            var vhost = a.hostname;
            var app = a.pathname.substring(1, a.pathname.lastIndexOf('/'));
            var stream = a.pathname.slice(a.pathname.lastIndexOf('/') + 1);

            // parse the vhost in the params of app, that srs supports.
            app = app.replace('...vhost...', '?vhost=');
            if (app.indexOf('?') >= 0) {
                var params = app.slice(app.indexOf('?'));
                app = app.slice(0, app.indexOf('?'));

                if (params.indexOf('vhost=') > 0) {
                    vhost = params.slice(params.indexOf('vhost=') + 'vhost='.length);
                    if (vhost.indexOf('&') > 0) {
                        vhost = vhost.slice(0, vhost.indexOf('&'));
                    }
                }
            }

            // when vhost equals to server, and server is ip,
            // the vhost is __defaultVhost__
            if (a.hostname === vhost) {
                var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
                if (re.test(a.hostname)) {
                    vhost = '__defaultVhost__';
                }
            }

            // parse the schema
            var schema = 'rtmp';
            if (url.indexOf('://') > 0) {
                schema = url.slice(0, url.indexOf('://'));
            }

            var port = a.port;
            if (!port) {
                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
                    port = url.indexOf(`webrtc://${a.host}:80`) === 0 ? 80 : 443;
                }

                // Guess by schema.
                if (schema === 'http') {
                    port = 80;
                } else if (schema === 'https') {
                    port = 443;
                } else if (schema === 'rtmp') {
                    port = 1935;
                }
            }

            var ret = {
                url: url,
                schema: schema,
                server: a.hostname,
                port: port,
                vhost: vhost,
                app: app,
                stream: stream,
            };
            self.__internal.fill_query(a.search, ret);

            // For webrtc API, we use 443 if page is https, or schema specified it.
            if (!ret.port) {
                if (schema === 'webrtc' || schema === 'rtc') {
                    if (ret.user_query.schema === 'https') {
                        ret.port = 443;
                    } else if (window.location.href.indexOf('https://') === 0) {
                        ret.port = 443;
                    } else {
                        // For WebRTC, SRS use 1985 as default API port.
                        ret.port = 1985;
                    }
                }
            }

            return ret;
        },
        fill_query: function (query_string, obj) {
            // pure user query object.
            obj.user_query = {};

            if (query_string.length === 0) {
                return;
            }

            // split again for angularjs.
            if (query_string.indexOf('?') >= 0) {
                query_string = query_string.split('?')[1];
            }

            var queries = query_string.split('&');
            for (var i = 0; i < queries.length; i++) {
                var elem = queries[i];

                var query = elem.split('=');
                obj[query[0]] = query[1];
                obj.user_query[query[0]] = query[1];
            }

            // alias domain for vhost.
            if (obj.domain) {
                obj.vhost = obj.domain;
            }
        },
    };

    self.pc = new RTCPeerConnection(null);

    // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
    self.stream = new MediaStream();

    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
    self.pc.ontrack = function (event) {
        if (self.ontrack) {
            self.ontrack(event);
        }
    };

    return self;
}

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher by WHIP.
function SrsRtcWhipWhepAsync() {
    var self = {};

    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
    self.constraints = {
        audio: true,
        video: {
            width: { ideal: 320, max: 576 },
        },
    };

    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
    // @url The WebRTC url to publish with, for example:
    //      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
    self.publish = async function (url) {
        if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);

        self.pc.addTransceiver('audio', { direction: 'sendonly' });
        self.pc.addTransceiver('video', { direction: 'sendonly' });

        if (
            !navigator.mediaDevices &&
            window.location.protocol === 'http:' &&
            window.location.hostname !== 'localhost'
        ) {
            throw new SrsError(
                'HttpsRequiredError',
                `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`,
            );
        }
        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);

        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
        stream.getTracks().forEach(function (track) {
            self.pc.addTrack(track);

            // Notify about local track when stream is ok.
            self.ontrack && self.ontrack({ track: track });
        });

        var offer = await self.pc.createOffer();
        await self.pc.setLocalDescription(offer);
        const answer = await new Promise(function (resolve, reject) {
            console.log('Generated offer: ', offer);

            const xhr = new XMLHttpRequest();
            xhr.onload = function () {
                if (xhr.readyState !== xhr.DONE) return;
                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                const data = xhr.responseText;
                console.log('Got answer: ', data);
                return data.code ? reject(xhr) : resolve(data);
            };
            xhr.open('POST', url, true);
            xhr.setRequestHeader('Content-type', 'application/sdp');
            xhr.send(offer.sdp);
        });
        await self.pc.setRemoteDescription(new RTCSessionDescription({ type: 'answer', sdp: answer }));

        return self.__internal.parseId(url, offer.sdp, answer);
    };

    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
    // @url The WebRTC url to play with, for example:
    //      http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
    self.play = async function (url) {
        if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1)
            throw new Error(`invalid WHEP url ${url}`);

        self.pc.addTransceiver('audio', { direction: 'recvonly' });
        self.pc.addTransceiver('video', { direction: 'recvonly' });

        var offer = await self.pc.createOffer();
        await self.pc.setLocalDescription(offer);
        const answer = await new Promise(function (resolve, reject) {
            console.log('Generated offer: ', offer);

            const xhr = new XMLHttpRequest();
            xhr.onload = function () {
                if (xhr.readyState !== xhr.DONE) return;
                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                const data = xhr.responseText;
                console.log('Got answer: ', data);
                return data.code ? reject(xhr) : resolve(data);
            };
            xhr.open('POST', url, true);
            xhr.setRequestHeader('Content-type', 'application/sdp');
            xhr.send(offer.sdp);
        });
        await self.pc.setRemoteDescription(new RTCSessionDescription({ type: 'answer', sdp: answer }));

        return self.__internal.parseId(url, offer.sdp, answer);
    };

    // Close the publisher.
    self.close = function () {
        self.pc && self.pc.close();
        self.pc = null;
    };

    // The callback when got local stream.
    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
    self.ontrack = function (event) {
        // Add track to stream of SDK.
        self.stream.addTrack(event.track);
    };

    self.pc = new RTCPeerConnection(null);

    // To keep api consistent between player and publisher.
    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
    // @see https://webrtc.org/getting-started/media-devices
    self.stream = new MediaStream();

    // Internal APIs.
    self.__internal = {
        parseId: (url, offer, answer) => {
            let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
            sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
            sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
            sessionid = sessionid.substr(0, sessionid.indexOf('\n'));

            const a = document.createElement('a');
            a.href = url;
            return {
                sessionid: sessionid, // Should be ice-ufrag of answer:offer.
                simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
            };
        },
    };

    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
    self.pc.ontrack = function (event) {
        if (self.ontrack) {
            self.ontrack(event);
        }
    };

    return self;
}

// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
function SrsRtcFormatSenders(senders, kind) {
    var codecs = [];
    senders.forEach(function (sender) {
        var params = sender.getParameters();
        params &&
            params.codecs &&
            params.codecs.forEach(function (c) {
                if (kind && sender.track.kind !== kind) {
                    return;
                }

                if (
                    c.mimeType.indexOf('/red') > 0 ||
                    c.mimeType.indexOf('/rtx') > 0 ||
                    c.mimeType.indexOf('/fec') > 0
                ) {
                    return;
                }

                var s = '';

                s += c.mimeType.replace('audio/', '').replace('video/', '');
                s += ', ' + c.clockRate + 'HZ';
                if (sender.track.kind === 'audio') {
                    s += ', channels: ' + c.channels;
                }
                s += ', pt: ' + c.payloadType;

                codecs.push(s);
            });
    });
    return codecs.join(', ');
}

export default {
    SrsRtcPublisherAsync,
    SrsRtcPlayerAsync,
    SrsRtcWhipWhepAsync,
    SrsRtcFormatSenders,
    SrsError,
};
